func_speex.c File Reference

Noise reduction and automatic gain control (AGC) More...

#include "asterisk.h"
#include <speex/speex_preprocess.h>
#include "asterisk/module.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/utils.h"
#include "asterisk/audiohook.h"

Include dependency graph for func_speex.c:

Go to the source code of this file.

Data Structures
struct	speex_direction_info
struct	speex_info

Macros
#define	DEFAULT_AGC_LEVEL   8000.0

Functions
static void	__reg_module (void)
static void	__unreg_module (void)
struct ast_module *	AST_MODULE_SELF_SYM (void)
static void	destroy_callback (void *data)
static int	load_module (void)
static int	speex_callback (struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
static int	speex_read (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
static int	speex_write (struct ast_channel *chan, const char *cmd, char *data, const char *value)
static int	unload_module (void)

Variables
static struct ast_module_info	__mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "Noise reduction and Automatic Gain Control (AGC)" , .key = ASTERISK_GPL_KEY , .buildopt_sum = AST_BUILDOPT_SUM, .load = load_module, .unload = unload_module, .load_pri = AST_MODPRI_DEFAULT, .support_level = AST_MODULE_SUPPORT_CORE, }
static struct ast_custom_function	agc_function
static const struct ast_module_info *	ast_module_info = &__mod_info
static struct ast_custom_function	denoise_function
static const struct ast_datastore_info	speex_datastore

Detailed Description

Noise reduction and automatic gain control (AGC)

Author

Brian Degenhardt bmd@d.nosp@m.igiu.nosp@m.m.com

Brett Bryant bbrya.nosp@m.nt@d.nosp@m.igium.nosp@m..com

The Speex 1.2 library - http://www.speex.org

Note

Requires the 1.2 version of the Speex library (which might not be what you find in Linux packages)

Definition in file func_speex.c.

Macro Definition Documentation

DEFAULT_AGC_LEVEL

#define DEFAULT_AGC_LEVEL   8000.0

Definition at line 49 of file func_speex.c.

Function Documentation

__reg_module()

static void __reg_module ( void  )

static

Definition at line 401 of file func_speex.c.

__unreg_module()

static void __unreg_module ( void  )

static

Definition at line 401 of file func_speex.c.

AST_MODULE_SELF_SYM()

struct ast_module * AST_MODULE_SELF_SYM

(

void

)

Definition at line 401 of file func_speex.c.

destroy_callback()

static void destroy_callback ( void *  data )

static

Definition at line 119 of file func_speex.c.

{
    struct speex_info *si = data;
 
    ast_audiohook_destroy(&si->audiohook);
 
    if (si->rx && si->rx->state) {
        speex_preprocess_state_destroy(si->rx->state);
    }
 
    if (si->tx && si->tx->state) {
        speex_preprocess_state_destroy(si->tx->state);
    }
 
    if (si->rx) {
        ast_free(si->rx);
    }
 
    if (si->tx) {
        ast_free(si->tx);
    }
 
    ast_free(data);
};

References ast_audiohook_destroy(), ast_free, speex_info::audiohook, speex_info::rx, speex_direction_info::state, and speex_info::tx.

load_module()

static int load_module ( void  )

static

Definition at line 387 of file func_speex.c.

{
    if (ast_custom_function_register(&agc_function)) {
        return AST_MODULE_LOAD_DECLINE;
    }
 
    if (ast_custom_function_register(&denoise_function)) {
        ast_custom_function_unregister(&agc_function);
        return AST_MODULE_LOAD_DECLINE;
    }
 
    return AST_MODULE_LOAD_SUCCESS;
}

References agc_function, ast_custom_function_register, ast_custom_function_unregister(), AST_MODULE_LOAD_DECLINE, AST_MODULE_LOAD_SUCCESS, and denoise_function.

speex_callback()

static int speex_callback ( struct ast_audiohook *  audiohook, struct ast_channel *  chan, struct ast_frame *  frame, enum ast_audiohook_direction  direction  )

static

Definition at line 149 of file func_speex.c.

{
    struct ast_datastore *datastore = NULL;
    struct speex_direction_info *sdi = NULL;
    struct speex_info *si = NULL;
    char source[80];
 
    /* If the audiohook is stopping it means the channel is shutting down.... but we let the datastore destroy take care of it */
    if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE || frame->frametype != AST_FRAME_VOICE) {
        return -1;
    }
 
    /* We are called with chan already locked */
    if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
        return -1;
    }
 
    si = datastore->data;
 
    sdi = (direction == AST_AUDIOHOOK_DIRECTION_READ) ? si->rx : si->tx;
 
    if (!sdi) {
        return -1;
    }
 
    if ((sdi->samples != frame->samples) || (ast_format_get_sample_rate(frame->subclass.format) != si->lastrate)) {
        si->lastrate = ast_format_get_sample_rate(frame->subclass.format);
        if (sdi->state) {
            speex_preprocess_state_destroy(sdi->state);
        }
 
        if (!(sdi->state = speex_preprocess_state_init((sdi->samples = frame->samples), si->lastrate))) {
            return -1;
        }
 
        speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_AGC, &sdi->agc);
 
        if (sdi->agc) {
            speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_AGC_LEVEL, &sdi->agclevel);
        }
 
        speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_DENOISE, &sdi->denoise);
    }
 
    speex_preprocess(sdi->state, frame->data.ptr, NULL);
    snprintf(source, sizeof(source), "%s/speex", frame->src);
    if (frame->mallocd & AST_MALLOCD_SRC) {
        ast_free((char *) frame->src);
    }
    frame->src = ast_strdup(source);
    frame->mallocd |= AST_MALLOCD_SRC;
 
    return 0;
}

References speex_direction_info::agc, speex_direction_info::agclevel, AST_AUDIOHOOK_DIRECTION_READ, AST_AUDIOHOOK_STATUS_DONE, ast_channel_datastore_find(), ast_format_get_sample_rate(), AST_FRAME_VOICE, ast_free, AST_MALLOCD_SRC, ast_strdup, speex_info::audiohook, ast_datastore::data, ast_frame::data, speex_direction_info::denoise, ast_frame_subclass::format, ast_frame::frametype, speex_info::lastrate, ast_frame::mallocd, NULL, ast_frame::ptr, speex_info::rx, speex_direction_info::samples, ast_frame::samples, speex_datastore, ast_frame::src, speex_direction_info::state, ast_audiohook::status, ast_frame::subclass, and speex_info::tx.

Referenced by speex_write().

speex_read()

static int speex_read ( struct ast_channel *  chan, const char *  cmd, char *  data, char *  buf, size_t  len  )

static

Definition at line 328 of file func_speex.c.

{
    struct ast_datastore *datastore = NULL;
    struct speex_info *si = NULL;
    struct speex_direction_info *sdi = NULL;
 
    if (!chan) {
        ast_log(LOG_ERROR, "%s cannot be used without a channel!\n", cmd);
        return -1;
    }
 
    ast_channel_lock(chan);
    if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
        ast_channel_unlock(chan);
        return -1;
    }
 
    si = datastore->data;
 
    if (!strcasecmp(data, "tx"))
        sdi = si->tx;
    else if (!strcasecmp(data, "rx"))
        sdi = si->rx;
    else {
        ast_channel_unlock(chan);
        ast_log(LOG_ERROR, "%s(%s) must either \"tx\" or \"rx\"\n", cmd, data);
        return -1;
    }
 
    if (!strcasecmp(cmd, "agc"))
        snprintf(buf, len, "%.01f", sdi ? sdi->agclevel : 0.0);
    else
        snprintf(buf, len, "%d", sdi ? sdi->denoise : 0);
 
    ast_channel_unlock(chan);
    return 0;
}

References speex_direction_info::agclevel, ast_channel_datastore_find(), ast_channel_lock, ast_channel_unlock, ast_log, buf, ast_datastore::data, speex_direction_info::denoise, len(), LOG_ERROR, NULL, speex_info::rx, speex_datastore, and speex_info::tx.

speex_write()

static int speex_write ( struct ast_channel *  chan, const char *  cmd, char *  data, const char *  value  )

static

Definition at line 204 of file func_speex.c.

{
    struct ast_datastore *datastore = NULL;
    struct speex_info *si = NULL;
    struct speex_direction_info **sdi = NULL;
    int is_new = 0;
 
    if (!chan) {
        ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
        return -1;
    }
 
    if (strcasecmp(data, "rx") && strcasecmp(data, "tx")) {
        ast_log(LOG_ERROR, "Invalid argument provided to the %s function\n", cmd);
        return -1;
    }
 
    ast_channel_lock(chan);
    if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
 
        if (!(datastore = ast_datastore_alloc(&speex_datastore, NULL))) {
            ast_channel_unlock(chan);
            return 0;
        }
 
        if (!(si = ast_calloc(1, sizeof(*si)))) {
            ast_channel_unlock(chan);
            ast_datastore_free(datastore);
            return 0;
        }
 
        ast_audiohook_init(&si->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "speex", AST_AUDIOHOOK_MANIPULATE_ALL_RATES);
        si->audiohook.manipulate_callback = speex_callback;
        si->lastrate = 8000;
        is_new = 1;
 
    } else {
        si = datastore->data;
    }
 
    if (!strcasecmp(data, "rx")) {
        sdi = &si->rx;
    } else {
        sdi = &si->tx;
    }
 
    if (!*sdi) {
        if (!(*sdi = ast_calloc(1, sizeof(**sdi)))) {
            if (is_new) {
                datastore->data = si;
                ast_channel_unlock(chan);
                ast_datastore_free(datastore);
            } else {
                ast_channel_unlock(chan);
            }
            return 0;
        }
        /* Right now, the audiohooks API will _only_ provide us 8 kHz slinear
         * audio.  When it supports 16 kHz (or any other sample rates, we will
         * have to take that into account here. */
        (*sdi)->samples = -1;
    }
 
    if (!strcasecmp(cmd, "agc")) {
        if (!sscanf(value, "%30f", &(*sdi)->agclevel))
            (*sdi)->agclevel = ast_true(value) ? DEFAULT_AGC_LEVEL : 0.0;
 
        if ((*sdi)->agclevel > 32768.0) {
            ast_log(LOG_WARNING, "AGC(%s)=%.01f is greater than 32768... setting to 32768 instead\n",
                    ((*sdi == si->rx) ? "rx" : "tx"), (*sdi)->agclevel);
            (*sdi)->agclevel = 32768.0;
        }
 
        (*sdi)->agc = !!((*sdi)->agclevel);
 
        if ((*sdi)->state) {
            speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_AGC, &(*sdi)->agc);
            if ((*sdi)->agc) {
                speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_AGC_LEVEL, &(*sdi)->agclevel);
            }
        }
    } else if (!strcasecmp(cmd, "denoise")) {
        (*sdi)->denoise = (ast_true(value) != 0);
 
        if ((*sdi)->state) {
            speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_DENOISE, &(*sdi)->denoise);
        }
    }
 
    if (!(*sdi)->agc && !(*sdi)->denoise) {
        if ((*sdi)->state)
            speex_preprocess_state_destroy((*sdi)->state);
 
        ast_free(*sdi);
        *sdi = NULL;
    }
 
    if (!si->rx && !si->tx) {
        if (is_new) {
            datastore->data = si;
            is_new = 0;
            ast_channel_unlock(chan);
        } else {
            ast_channel_datastore_remove(chan, datastore);
            ast_audiohook_remove(chan, &si->audiohook);
            ast_channel_unlock(chan);
            ast_audiohook_detach(&si->audiohook);
        }
        ast_datastore_free(datastore);
        return 0;
    }
 
    if (is_new) {
        datastore->data = si;
        ast_channel_datastore_add(chan, datastore);
        ast_channel_unlock(chan);
        ast_audiohook_attach(chan, &si->audiohook);
    } else {
        ast_channel_unlock(chan);
    }
 
    return 0;
}

References ast_audiohook_attach(), ast_audiohook_detach(), ast_audiohook_init(), AST_AUDIOHOOK_MANIPULATE_ALL_RATES, ast_audiohook_remove(), AST_AUDIOHOOK_TYPE_MANIPULATE, ast_calloc, ast_channel_datastore_add(), ast_channel_datastore_find(), ast_channel_datastore_remove(), ast_channel_lock, ast_channel_unlock, ast_datastore_alloc, ast_datastore_free(), ast_free, ast_log, ast_true(), speex_info::audiohook, ast_datastore::data, DEFAULT_AGC_LEVEL, speex_info::lastrate, LOG_ERROR, LOG_WARNING, ast_audiohook::manipulate_callback, NULL, speex_info::rx, speex_callback(), speex_datastore, speex_info::tx, and value.

unload_module()

static int unload_module ( void  )

static

Definition at line 380 of file func_speex.c.

{
    ast_custom_function_unregister(&agc_function);
    ast_custom_function_unregister(&denoise_function);
    return 0;
}

References agc_function, ast_custom_function_unregister(), and denoise_function.

Variable Documentation

__mod_info

struct ast_module_info __mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "Noise reduction and Automatic Gain Control (AGC)" , .key = ASTERISK_GPL_KEY , .buildopt_sum = AST_BUILDOPT_SUM, .load = load_module, .unload = unload_module, .load_pri = AST_MODPRI_DEFAULT, .support_level = AST_MODULE_SUPPORT_CORE, }

static

Definition at line 401 of file func_speex.c.

agc_function

struct ast_custom_function agc_function

static

Definition at line 366 of file func_speex.c.

                                                 {
    .name = "AGC",
    .write = speex_write,
    .read = speex_read,
    .read_max = 22,
};

Referenced by load_module(), and unload_module().

ast_module_info

const struct ast_module_info* ast_module_info = &__mod_info

static

Definition at line 401 of file func_speex.c.

denoise_function

struct ast_custom_function denoise_function

static

Definition at line 373 of file func_speex.c.

                                                     {
    .name = "DENOISE",
    .write = speex_write,
    .read = speex_read,
    .read_max = 22,
};

Referenced by load_module(), and unload_module().

speex_datastore

const struct ast_datastore_info speex_datastore

static

Initial value:

= {
    .type = "speex",
    .destroy = destroy_callback
}

Definition at line 144 of file func_speex.c.

                                                         {
    .type = "speex",
    .destroy = destroy_callback
};

Referenced by speex_callback(), speex_read(), and speex_write().